1. Field of the Invention
The present invention relates to a voice over Internet protocol-Unified Messaging System (VoIP-UMS) accessed via a private branch exchange or an IP-phone using a Session Initiation Protocol (SIP) and, more particularly, to a VoIP-UMS and method of accessing the same in which an initial call setup message is sent which includes an INFO message representing call information when the VoIP-UMS is accessed for the purpose of using functions, such as voice mail or the like.
2. Description of the Related Art
A Voice over Internet Protocol (VoIP) refers to Internet Protocol (IP) telephone technology for a series of equipment that delivers voice information using IP. Generally, the VoIP is not a traditional protocol based on a circuit like PSTN, Public Switched Telephone Network. The VoIP sends the voice information within discontinuous packets in a digital form.
A primary advantage of the VoIP and Internet telephony technology is that telephone users can get long-distance and international telephone services under Internet and Intranet environments only with local phone charges because an integrated telephone service is implemented by utilizing an existing IP network as it is.
For a voice call through the VoIP, using a public network cannot guarantee the same quality of service (QoS) as that in a circuit network, and thus, a private network, managed by a personal company or an Internet telephone service provider, may be used for a high quality of service.
The Internet telephone service provider is operating a server that manages an IP address corresponding to a called party's telephone number, and allowing a user to make a voice call over the Internet without separately managing the IP address corresponding to the called party's telephone number.
The VoIP may allow a customer to directly talk with a staff or a consultant so long as the customer presses a button on the Internet while navigating the Internet without using a separate telephone. A call center to which the VoIP has been applied is configured into two types of systems. The one is of a private automatic branch exchange (PABX) standalone type having a separate VoIP gateway and the other is of a private automatic branch exchange (PABX) integrated type in which a PABX has a VoIP gateway included therein.
To use the VoIP, a company must have a gateway with a VoIP equipment disposed therein. The gateway receives a delivered voice, which has been segmented into packets, from users in the company, and forwards the packets to other parts on the Intranet or to the PSTN using a T-1 (A dedicated digital connection supporting data rates of 1.544 Mbits per second, 24 channels at 64 Kbps, also called DS 1) or E1 (European format for digital transmission, carrying signals at 2 Mbps, 32 channels at 64 Kbps, with 2 channels reserved for signaling and controlling) interface.
The VoIP uses a real-time transport protocol (RTP) in order to support that packets arrive at on time, in addition to its original IP function. Using the public network can hardly guarantee quality of service (QoS). Using any private network managed by the personal company or the Internet telephone service provider (ITSP) may provide a more excellent service.
A Unified Messaging System (UMS) refers to a system that stores and manages various types of all messages, such as voice, Fax (facsimile), E-mail (electronic-mail), and the like, in one logical post-box. The messages are accessible to PCs (personal computers) as well as a variety of communication media, such as telephones, Faxes, mobile phones, or the like. A user suffices to have only one kind of interface regardless of the types of the message because different types of messages such as voice mails, Faxes, E-mails or the like are allowed to be retrieved, formulated and exchanged in one mail box by unifying wired and wireless telephones and data communication networks.
The UMS is more advanced technology based on existing E-mail/VMS technology. It has been extensively applied to several fields such as call centers, companies, ISP businesses, special category communications, mobile communications, or the like by enabling rapid, exact and smooth communications to be performed between an individual and a company and between a company and a company.
The voice communication technology and the data communication technology have been developed based on different technologies. However, with the advent of VoIP technology which digitalizes data to be handled as text information, discrimination between the voice communication and the data communication becomes ambiguous.
The trend of unifying both voice and data leads to UMS that is a solution that provides effective management on numerous messages regardless of the types of the messages, and provides the bi-directional use.
The development and rapid distribution of the Internet technology are shifting a communication service basis to the Internet. As markets such as companies, communication businesses or the like are changed rapidly, users highly desire to handle several types of incoming information in an integrated manner. A system satisfying this desire is UMS called next generation CTI technology.
A Session Initiation Protocol (SIP) is a standard for VoIP connection setup. The SIP is an application layer control protocol that sets, modifies and terminates a session in a client-server manner and is based on a very simple text.
The SIP is not fixed to any protocol stack but is easy to extend and use because it is based on a text such as HTTP (hypertext transfer protocol). The SIP is composed of a user agent and a network server. The user agent is a termination system and is composed of a user agent client and a user agent server.
The SIP operates in such a transaction processing method that a client sends a service request message to a server, and the server processes it and then sends a response message to the client, like HTTP.
Users that communicate using the SIP will use a Uniform Resource Identifier (URI) of a “user@host-plus-domain” format, similar to an E-mail address, as an identifier between the respective users.
Since the SIP is based on IP, it does not have to consider inter-operability as in H.323. The SIP is simpler than H.323, thereby implementing at a low cost. In addition, an instruction or message format is easily decoded or debugged and extended because it is simple. Security, management extensibility or the like is excellent due to a client-server structure.
The SIP provides two connection methods to a called party, i.e., via a proxy server and directly. A calling party will request a call to a called party by sending an INVITE message using an addressing manner suitable for the SIP.
Addresses used in the SIP can be largely classified into three addresses. First, the combination of a network name and a host name used by a user with an Id (identification) used for the user to login may be used as the SIP address. Second, an IP address itself may be used as the SIP address. Third, an E-mail address or a DNS (domain name system, server, or service) name used in an existing homepage may be used as the SIP address. Using an existing E-mail address is the most realized manner.
Call setup is classified into several methods according to situations.
First, determination is made regarding whether the call setup to a called party is attempted via a proxy server or directly. If a calling party knows a called party's address and is allowed to directly set up a call to the called party, the calling party will set up a call directly. On the other hand, in the case where the call setup is made via the proxy, the calling party will set up a call after discovering the called party's address with reference to a DNS lookup table.
A second method is to discover a called party's address using a request response protocol and to set up a call. A request message will be delivered using a predefined, well known port number through TCP (transmission control protocol) or UDP (user datagram protocol).
If a client user agent receives a message via the above-defined, well known port while the user examines the network, the agent will send a response message to participate in telephone or video conference. If a problem associated with the received message or the transmitted message occurs due to the user agent or the server, the agent sends a message indicating that an ICMP (Internet control message protocol) message is not allowed to arrive, and notifies that the problem occurs.
All SIP messages are based on a text. Upon delivering the messages, several ones of the messages are sent by one TCP segment or UDP datagram using TCP or UDP.
The SIP has been developed in consideration of scalability, extensibility, flexibility, interoperability, and the like. Accordingly, the SIP has advantages that the SIP is more concise than competitive H.323, is easily integrated and interworked with an existing Internet/web environment, and is easy to enhance and extend. Using the SIP enables various types of new multimedia communication services on the Internet, such as a VoIP service, to be developed with relatively low cost and less time.
Particularly, the introduction of the SIP can easily solve insufficient interworking between VoIP business networks, incomplete interoperability between VoIP equipment, or the like, which are regarded as the greatest weaknesses in H.323.
Thus, a user is able to use a multimedia communications service throughout the world only with one SIP URI (uniform resource identifier) allocated to the user, thereby providing the maximized convenience and work efficiency to the user.
The advent of the SIP leads to many spreading effects in communication service markets using the Internet. Most of the existing VoIP systems have been implemented based on an H.323 protocol that is adopted as a standard by ITU-T (International Telecommunication Union-Telecommunication standardization sector).
However, since H.323 is a technical scheme that is originally developed to allow for multipoint voice, video, and data communications on a packet-switching LAN network, it basically has limitations in supporting a broadband network and a great number of users.
Internet telephone technologies have been highly spotlighted as marketable technology because VoIP related markets have become larger. At this time, SIP that is a signaling protocol for bi-point/multi-point communications on Internet becomes paid attention as technology substituting for an existing H.323.
The SIP is simpler than H.323. In SIP, if a caller party sends to a called party an invite message including information on the caller party and session information for exchanging multimedia data, then the called party notifies whether to accept it. SIP has a connection process simpler than that of H.323 while SIP has a disadvantage in that it cannot recognize all capabilities of the terminal. SIP has a regulation on inter-communication between servers while H.323 has no regulation.
Currently, The SIP is being used or developed as a call signaling protocol for application services in several fields. As previously described, SIP is a text based protocol such as HTTP, and employs the same identifier having an address system similar to an E-mail address system so that a voice call service as well as an E-mail service, an instant message service, or the like is provided without regard to when and where.
The SIP supports a capability-based service depending on use capability through a session parameter within a body portion of an SIP message format when a session is established, the SIP message format being composed of a head and a body.
The SIP users can register their own portable phone numbers, office numbers, home telephone numbers, E-mail addresses, or the like in a server. In addition, processing contents or the like for all calls can be stored in the server. Since the SIP is a text-based protocol, the type of message is a request/response format composed of a method and a response to the method.
Meanwhile, to allow the VoIP-UMS to be used with the SIP, message delivery may be performed through a Private Branch eXchange (PBX).
Normally, a private branch exchange or a key phone system (hereinafter, referred to as ‘private branch exchange’) refers to an in-plant switching equipment disposed in plants such as government and public offices, companies, hospitals, or the like.
The private branch exchange includes an extension subscriber card (SLIC: Subscriber Line Interface Card) accommodating extension subscribers, to which extension telephones are connected, and a trunk card connected to a central office line (COL) that is connected to a central office exchange. At this time, a number of extension subscribers may make calls therebetween without passing through an external central office line, and may make an outgoing call to the external central office line after dialing a trunk access code, such as ‘9’ and then an external telephone number.
Generally, extension lines for extension subscribers, connected to the private branch exchange, include extension lines accommodating typical analog telephones, key phone lines accommodating key phone telephones, ISDN Basic Rate Interface (ISDN BRI) lines accommodating Integrated Services Digital Network (ISDN) telephones, lines accommodating multi-function telephones (e.g., digital telephones), and the like. These lines are each connected to a back board of a matching device mounted on the private branch exchange in a board form.
Further, the central office lines connected to the private branch exchange include an analog trunk, a digital trunk such as an E1 line, a T1 line or the like, an ISDN Primary Rate Interface (ISDN PRI) line, and the like. These lines are connected to the back board of the matching device mounted on the private branch exchange.
Using VoIP in the private branch exchange requires a gateway. This gateway serves to receive voice data, which is segmented into packets and then transmitted, from users and deliver the data to a destination over a network such as Internet, Intranet or the like, or to directly connect a relevant call to PSTN using an analog trunk and a T1 or E1 interface.
Normally, upon using the VoIP gateway connected to the private branch exchange, the private branch exchange separates a central office line connected to the VoIP gateway and offers a different access code to the central office line.
For example, when an extension subscriber desires to make a voice call with an external called party via the VoIP gateway, a trunk access code is allocated to ‘8’ while when the extension subscriber desires to make a call with an external called party via a typical central office line connected to PSTN, a trunk access code is allocated to ‘9’. Thus, the divided codes are used.
Further, in the case where a desired voice call is made via the VoIP gateway, only an outgoing call is permitted. An incoming call is permitted via the central office line.
To make a VoIP call, first, a user should pick up a telephone receiver, confirm a dial tone from the private branch exchange, and then press a dial number to connect to a VoIP gateway that serves to connect the private branch exchange and TCP/IP network (Internet).
At this time, the VoIP gateway will inquire a routing table to see whether the entered number is a serviceable number.
If the entered number is not the serviceable number, the VoIP gateway confirms whether connection to another VoIP gateway is required. If not required, the gateway returns the relevant information to the private branch exchange to induce a call attempt via a typical telephone network.
If the VoIP gateway discovers an Internet path corresponding to the entered number, it allows a call to be made. For this purpose, the gateway should secure a line to a VoIP gateway of a called party.
The VoIP gateway at a calling party modulates voice into IP packets and then transmits relevant IP packets via a given path over the TCP/IP network as if they are relevant data packets.
The VoIP gateway at a receiving side receiving the IP packet data recombines packet information to restore into an analog signal, route the restored signal in a phone call form via PSTN in the exchange station or another private branch exchange, and directly connects a call to the receiving telephone. Thus, the routing procedure for a voice call via Internet is completed.
Meanwhile, in the earlier art, when a user accesses the UMS, the SIP does not send, on an initial call setup message, information for the currently attempted call, for example, call information such as NoAnswer Forward, Busy Forward or the like, but sends the information using a separate message. Accordingly, the UMS cannot recognize call information until receiving a message containing the call information, which obstructing rapid response. Further, there was a problem that message loss, delay, or the like is caused due to the property of the VoIP network, resulting in difficulty in function and implementation.